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Entries in ISDN (16)

Monday
Apr222013

The new Skype: The latest Source Connect rival?

Skype has recently released a new codec to their system that improves the sound quality of audio transmissions. What does that mean to you? It means that now the audio quality that you can achieve via Skype is starting to rival the audio quality that can be sent via other much more expensive systems such as ISDN and Source Connect. The question:  Is it really good enough for professional studio use or is it just close but no cigar?

I decided to put this new Skype version to the test. First, how do you know if you have the new Skype version? Well, from what I can tell, this updated audio codec called Opus is working in the latest updates (read comment below).  I'm not aware of what exact version of Skype you have to have installed for this new codec to start working but on my Mac as of this writing I'm running version 6.3.0.602. The audio quality really is quite remarkable as well as the latency is very low. That means that you can have a conversation via Skype and almost never step on each other's words because you're not waiting for the other person's transmission to reach you, making it much easier to communicate in real time. Anybody that has ever worked by ISDN or Source Connect knows what I'm talking about.  

So let's first talk about how Skype's new audio quality rivals Source Connect. For those who are not familiar with Source Connect, it is a professional audio software from Source Elements that allows you to stream studio quality audio via internet connection. It's been available for at least 5 years now and has  pretty strong foothold in the voiceover industry. Of all of the competing systems that are becoming available, Source Connect definitely has the biggest head start. Source Connect is cross-platform so it will run on Mac or Windows and but is not available yet to run on iOS or Android phones or tablets.

Most of you are likely familiar with Skype as it allows audio transmission via the internet for free. It's very flexible because it allows video, screen sharing, calls to landlines, chat and to share files with the other user or users who you are connected with during the call. Skype does run on iPhone, iPad, Android, Windows, Mac and Linux among others. Both Skype and Source Connect can be used to communicate in real time time with very little latency with another studio and sound quality is quite remarkable.

However, when we start looking at the resulting recording made via Skype or Source Connect you start to notice the differences between the two systems. My initial tests show that while the audio quality from Skype is surprisingly good, especially considering the price, it does fall short in regards to pure audio quality. Some portion of the Skype codec, the way that Skype handles its audio, is adding dynamic compression to the signal. The resulting recorded audio file clearly has been limited or compressed and some way by Skype. This could be a problem depending on the needs of the studio who is receiving your file. If they want your audio to be completely in it's original state with no processing of any kind, Skype will not be an acceptable substitute for Source Connect.

Take a listen to an audio sample I recorded of voice actor Graeme Spicer with SC then Skype, or download the WAV file for further analysis. 

 

I think you will agree that the quality would acceptable for radio spots, field reporting, and many projects where immediate access to remote talent is required.  Skype will make an excellent backup to SC while traveling in areas with poor broadband Internet access or network firewall issues preventing a two way connection via SC.

I also spoke about this topic during the first segment of EWABS Episode 91, which you can watch here

 

 

Wednesday
Apr102013

VO STUDIO: A show focusing on VO's home studios

I got to consult on Kami and Kim's episodes, but I appear in Kami's episode.  

If you like the video, please Subscribe on Youtube and Share on Facebook!

 

Tuesday
Jan082013

In Or Outside The Box: Is it the Codec or the Network? by Dave Immer


I have noticed in some blogs a tendency to intermingle codec problems with network quirks.

For instance, Source-Connect is an excellent piece of software that delivers very good audio using the AAC algorithm. When the network over which it is running is up to the task, results are satisfying. But a user expecting to get an “ISDN experience” from it when their internet service is experiencing packet collisions, jitter and bursty or slow performance, may have a tendency to associate such behavior with the software codec itself when that is not the case.

Another example would be the Telos Zephyr. Being as the Zephyr is immensely popular and has emerged as the standard ISDN codec (in the US,) people tend to co-mingle it’s behavior in their minds with the ISDN line to which it is connected. While the Zephyr is a well designed, user friendly box, it can be unstable compared to other major brands. The characteristics of such a “standard” codec might be perceived as the nature of ISDN itself, which, again, is not necessarily so.

Certainly troubleshooting is inevitable with any unit. And unless you have the option of substituting alternate equipment, networks or software, it’s way harder to arrive at a clear understanding of the problem(s.) So:
1. Rent one or ask a colleague if you can borrow their codec and try it on your line.
2. Take your codec over to a colleague’s network and try it on their line.
3. Involve a third party like the Digifon Bridge to make test connections. 
This way you can at least determine if the issue is in or outside the box (or both!)

Let me know if you have comments about this.                   Thanks,

Dave                             Complete library of newsletters:  www.digifon.com/blog.html

Friday
Jan042013

IP Codecs: Main or Backup/Alternate? by Dave Immer


If an IP codec is your main live audio networking tool, then it’s probably because
A. You can’t get ISDN or are unwilling to pay for it,
B. Most of your clients use IP anyway.
But if ISDN is your primary codec, you should definitely also have an IP codec. This gives you security as a backup, plus flexibility to connect with compatible IP-only systems. And everybody can get IP.

With ISDN MPEG codecs from companies such as Telos, Musicam, APT, Mayah, Prodys, TieLine, AudioTX, etc, compatibility is not (usually) an issue, as manufacturers long ago agreed on connection standards. But with IP codecs, it’s still kind of like the wild west with Source-Connect on it’s proprietary (albeit popular) mountain, and most other IP codec companies providing limited cross-compatibility despite employing standard algorithms such as AAC and MPEG layers II & III.

No matter what IP codec you use, the usual internet problems will persist such as latency and uncertain reliability. But these disadvantages can be minimized with the right algorithms on the right hardware platform over the right network. My recommendation for IP codecs:

1. Algorithm: APTX or AAC Low Delay
2. Hardware Platform: Musicam Suprima family or Source-Connect on a fast Intel-based processor.
3. Network: 15Mb/5Mb cable/fios/u-verse

Let me know if you have another IP codec you particularly like.                    Thanks,

Dave                             Complete library of newsletters:  www.digifon.com/blog.html

Thursday
Nov152012

WAKE UP! (Your ISDN Codec, That Is...) by Dave Immer

Both you and the remote facility are configured identical, the calls connect, but only one side is locked (framed). But when audio signal gets sent to the un-locked end, it becomes locked. If you own an older ISDN codec model such as the Telos Zephyr (both Classic & Xstream) or the Musicam Prima (both CDQ & LT) you may, from time to time, need to receive audio to ‘wake up the box.’

Of course, the user that needs to send the audio signal to wake up the other end doesn’t know the other end is not locked, since the locked (framed) condition only pertains to the decoder (receive.) So it’s good to send some signal as soon as the dial-up connection is made to make sure the remote side codec is ‘awake.’

I have also seen phantom signal appear on codec meters looking like a steady noise level, only to go away once an ISDN connection is established.

Often the equivalent of jiggling the wires or toggling the switch clears up the issue.

Sometimes a reboot is needed to get stuff to work. (The display on the Zephyr may show ‘Ready/Ready’ but the box cannot make or receive calls.) I put this in the same category as wire jiggling since no settings get changed to remedy the condition.

Let me know if you have questions about waking up the box.                  Thanks,

-Dave                     Complete library of newsletters:  www.digifon.com/blog.html